1. Field of the Invention
The present invention relates to a sound volume control unit and a method for automatically adjusting sound volume of sound signals.
2. Description of the Related Art
A multi-point speech system for speaking among two or more points by using cellular phones, IP (Internet Protocol) telephones, television conference systems and the like has come to be used lately. In such multi-point speech system, sound volume of receiving signals may differ per point due to sensitivity of microphones of transmitting-side units of the respective points even if reproducing volume of a receiving-side unit is set at certain level.
FIG. 19 shows an exemplary configuration of such multi-point speech system. Transmitting-side telephones 101 and 102 as well as a receiving-side telephone 104 are connected to a communication network 103. The telephone 104 receives a voice signal S1 from the telephone 101 and a voice signal S2 from the telephone 102 and a speaker 105 converts output signals and outputs as voice.
When volume of the voice signal S1 is large and volume of the voice signal S2 is small at this time, volume of the voices outputted out of the speaker 105 differs between those from the telephone 101 and the telephone 102. Then, it has been desired to automatically adjust the volumes of the receiving signals in order to make the volumes of sounds of the all points even.
FIG. 20 shows a prior art sound volume control method using automatic gain controls (AGC). The AGC is a function for automatically adjusting an amplification factor (gain) of an amplifying circuit so that volume of an output is adjusted to a desirable level even when amplitude of an input signal fluctuates.
A mixer 203 performs mixing (addition) in this volume control method after leveling the volumes of the receiving signals S1 and S2 of the respective points by the AGCs 201 and 202, respectively. Thereby, it becomes possible to correct the difference between the volumes of the points. Various configurations have been proposed as the configuration of the AGCs 201 and 202.
FIG. 21 is a structural view of an AGC described in a non-patent literature, Peter L. Chu, “VOICE-ACTIVATED AGC FOR TELECONFERENCING” proceedings ICASSP96 vol. 2, pp. 929-932, 1996. According to this configuration, a frame electric power calculating section 301 divides an input signal into frames of 20 ms and calculates energy (frame power) within each frame. Next, a maximum value calculating section 302 calculates a maximum value of the frame power from the past to the present time and a gain calculating section 303 calculates a gain from a difference of powers between the maximum value and a target level. Then, a multiplier 304 multiplies the gain with the input signal to generate an output signal.
However, although the volumes of voices of speakers in the output signals of the AGC are almost leveled in all of the points, volume of noise that depends on an ambient environment differs per each point. Still more, a SNR (Signal-to-Noise Ratio) of each point does not change. Accordingly, a SNR of an output signal after mixing is adjusted to a value of a point where the SNR of the receiving signal is least among all of the points. Therefore, when there is such point where the SNR is small, the SNR of all of the points becomes small and it becomes hard to catch the voices.
Japanese Patent Application Laid-open No. 2004-133403 relates to a voice signal processing apparatus that samples voices that form a conversation in a conversation state in which a plurality of voices and noises are mixed and raises an output volume of its voice or lowers a volume of other sounds.
Japanese Patent Application Laid-open No. 2004-507141 relates to a method for processing a voice signal to overcome background noise not related to the voice signal, Japanese Patent Application Laid-open No. 2002-223268 relates to a voice control unit for obtaining receiving voice from which a discomfort feeling is eliminated without being buried in an ambient background noise and Japanese Patent Application Laid-open No. 2002-1575100 relates to an adaptive noise suppressing voice coding apparatus that detects and eliminates noises within a present speech.